一番最新のCisco 300-815試験問題集PDFには2023年更新
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Cisco 300-815 認定試験の準備をするために、候補者は Cisco の公式トレーニングコース、Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) を受講することができます。このコースは、試験の目的をすべてカバーしています。候補者は、練習問題、学習ガイド、オンラインビデオなどの他の学習教材を利用して、学習を補完することもできます。
Cisco 300-815 認定試験は、Cisco 環境における高度なコールコントロールおよびモビリティサービスの実装能力をネットワークプロフェッショナルに求める試験です。この試験は、ネットワーキング分野でキャリアを追求する人や、既に業界で活躍している人がスキルや知識を向上させたい場合に最適です。また、ネットワークプロフェッショナルが自分の専門知識を雇用主や同僚に証明するための素晴らしい方法でもあります。
質問 # 58
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be associated under the route group
- B. can be associated only under the route list
- C. chooses the route group that is configured under the device pool of the calling-party device
- D. chooses the route group that is configured under the device pool of the called-party device
- E. can be assigned directly to the route pattern
正解：B、C
質問 # 59 
Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user
C. Which two scenarios are correct? (Choose two.)
- A. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.
- B. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.
- C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
- D. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- E. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
正解：A、D
解説：
Section: Signaling and Media Protocols
質問 # 60
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
- A. interworking between h245-signal and rtp-nte
- B. interworking between an OOB method and RFC2833 for flow-through calls
- C. interworking between h245-alpha numeric and sip-kpml
- D. interworking between an OOB method and RFC2833 for flow-around calls
正解：D
解説：
Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35
質問 # 61
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?
- A. debug H.323 asn 1
- B. debug H.225 asn1
- C. debug H.225 media
- D. debug H.323 messages
- E. debug H.246 asn 1
正解：B
解説：
Section: Signaling and Media Protocols
質問 # 62
An administrator is asked to configure egress call routing by applying globalization and localization on Cisco UCM. How should this be accomplished?
- A. Globalize the calling and called numbers to PSTN format and localize the calling number in the gateway.
- B. Localize the calling and called numbers to E. 164 format and globalize the called number in the gateway.
- C. Localize the calling and called numbers to PSTN format and globalize the calling and called numbers in the gateway.
- D. Globalize the calling and called numbers to E. 164 format and localize the called number in the gateway.
正解：D
質問 # 63
Refer to the exhibit.
An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu. Which two steps must be performed to resolve this issue? (Choose two.)
- A. Activate the Cisco Dialed Number Analyzer service.
- B. Activate the Cisco Extended Functions service.
- C. Activate the Cisco CallManager service.
- D. Activate the Cisco Dialed Number Analyzer Server service.
- E. Restart the subscriber
正解：A、D
質問 # 64
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
- A. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
- B. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
- C. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- D. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
正解：A
解説：
Reference:
https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified- cm-using-rtmt/ta-p/3161200
質問 # 65 
Refer to the exhibit. An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu.
Which two steps must be performed to resolve this issue? (Choose two.)
- A. Activate the Cisco Dialed Number Analyzer service.
- B. Activate the Cisco Extended Functions service.
- C. Activate the Cisco CallManager service.
- D. Activate the Cisco Dialed Number Analyzer Server service.
- E. Restart the subscriber
正解：A、D
解説：
Section: Call Control and Dial Planning
質問 # 66
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. OS Administration Page
- B. MTP
- C. CCSIP
- D. RTMT
正解：D
解説：
Section: Call Control and Dial Planning
質問 # 67
Refer to the exhibit.
ILS has been configured between two hubs using this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?
- A. A password has never been set for ILS.
- B. Trust certificates for ILS have not been installed on the clusters
- C. Use TLS Certificates must be selected.
- D. The Cluster IDs have not been set to unique values
正解：D
質問 # 68
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
- A. o= line of SDP content
- B. c= line of SDP content
- C. Allow: header if the 200 OK response
- D. Contact: header of the 200 OK response
正解：B
質問 # 69
An IP Telephony administrator is deploying IP phones The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111 2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two )
- A. Option B
- B. Option D
- C. Option E
- D. Option A
- E. Option C
正解：B、E
質問 # 70
Refer to the exhibit.
An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?
- A. SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1 - B. SIP binding in SIP configuration mode:
voice service voip sip
bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0 - C. SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0 - D. SIP binding In dial-peer configuration mode:
dial-peer voice 300 voip
voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1
正解：C
質問 # 71
Which IOS command creates a SIP-enabled dial peer?
- A. dial peer voice 20 sip
- B. voice dial-peer 20 sip
- C. dial-peer voice 20 voip
- D. dial-peer voice 20 pots
正解：C
解説：
Reference:
https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6
質問 # 72
Refer to the exhibit.
Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?
- A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- B. No DTMF is negotiated.
- C. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
- D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
正解：B
質問 # 73
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Configure the command no ip address trusted authenticate under "voice service voip".
- B. Enable Secondary Dial tone on Analog and Digital FXO Ports.
- C. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
正解：C
解説：
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/ cmeadm/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2
質問 # 74
A user in location X dials an extension at location
Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?
- A. codec mismatch
- B. ptime mismatch
- C. phone class of service issue
- D. missing Call Admission Control
正解：A
解説：
Section: Cisco Unified Border Element
質問 # 75
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
- A. c= line of SDP content
- B. o= line of SDP content
- C. Allow: header if the 200 OK response
- D. Contact: header of the 200 OK response
正解：B
解説：
Section: Signaling and Media Protocols
質問 # 76
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. ALERTING
- B. CONNECT
- C. RINGING
- D. PROCEEDING
正解：B
質問 # 77
What are the elements for Device Mobility configuration?
- A. physical location. Device Mobility group, and region
- B. device pool, Device Mobility group, and region
- C. physical location, device pool, and Device Mobility group
- D. device pool, Device Mobility group, and Cisco IP phone
正解：C
質問 # 78
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. target carrier-id
- B. incoming called-number
- C. incoming uri
- D. answer-address
正解：C
解説：
Reference:
https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8
質問 # 79
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
- A. *.*
- B. *@example.com
- C. example.com
- D. [email protected]
- E. gateway.corporate.local
正解：A、B
質問 # 80
Refer to the exhibit.
In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C.
Which two scenarios are correct? (Choose two.)
- A. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
- B. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.
- C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.
- D. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- E. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
正解：A、D
質問 # 81
......
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